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    12,760 application android sip voip робіт знайдено, ціни вказані в USD

    Що потрібно… «Розвернути» на сервері ASTERISK Налаштувати транк sip gsm +transfer Запис розмов та фіксацію номерів трансферних Адмін панель Панель доступу user Надаштування GSM шлюзів

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    Gsm voip gatway 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    I need an Android app. I would like it designed And Build Android application make mobile phone act as GSM to voip gateway To connect with Asterisk PBX We needs to use an android cellphone as an asterisk channel/Gateway. His aim is to make phone calls using his Asterisk PBX through a cellphone running android. It aim to make phone calls using

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    Country Sales Manager UK 4 дні(-в) left
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    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a

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    Sales Representatives for SaaS in the UK 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

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    VoIP telephony 3 дні(-в) left
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    We have iPECS-100 behind Chekpoint firewall. All customers use soft phones clients. Have periodically experience communication problems. Need a person for a one-time or permanent job

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    LinPhone Development Project 3 дні(-в) left
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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    Low bandwidth voip solution by asterisk. 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server

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    I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

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    android voip call application 3 дні(-в) left
    ПІДТВЕРДЖЕНО

    System requirements Registra...and database mysql Through the application you can make a voice call between registered users on the system The user is activated after registration through the database The IP number, MAC address, device type and user name are stored in the database The call is secure and encrypted between the parties Simple application

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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

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    Місцевий
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    Freepbx and Asterisk settings 2 дні(-в) left
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    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

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    Personalisation LinPhone Security & Design 1 день left
    ПІДТВЕРДЖЕНО

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    Project for Anton F. 4 дні(-в) left

    ...I have a IP PBX with 2 sets of 10 numbers. Set nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my

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    JSP JAVA expart Need FOR VOS3000 VOIP LINUX Server 1 день left
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    i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

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    Customization of LinPhone SIP / VoIP Client 23 годин(-и) left
    ПІДТВЕРДЖЕНО

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    10 заявки
    VOIP yate call pass 22 годин(-и) left
    ПІДТВЕРДЖЕНО

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    app development 21 годин(-и) left

    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

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    VOIP phone system 20 годин(-и) left
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    we need a phone system based on twilio platform.

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    Wordpress Plugin for Phone Call 15 годин(-и) left
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    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

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    Actionscript 3.0 SIP 10 годин(-и) left
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    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

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    PHP SIP client 8 годин(-и) left
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    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    VOIP Project 4 годин(-и) left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

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    Місцевий
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    Create SNOM Phone Dial Plan Закінчився left

    I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [увійдіть, щоб побачити URL] or yate or besip [увійдіть, щоб побачити URL]

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    Integrazione api voip Закінчився left

    Vorrei integrare un sistema VoIP nel mio sito web così cliente possono vedere la loro fattura e possiamo vedere i loro menù.

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    add sip trunk to elastix Закінчився left

    add sip trunk to elastix i have sip trunk from STC

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    Paid VOIP services app Закінчився left

    Application that allows people to instantly phone a specialist, on a pay-per-minute basis, to receive recommendations

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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...

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    ...installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Transfer &bu...

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    Raspberry pi as VoIP GSM gateway Закінчився left

    I want to use raspi as gsm gateway. This is the basic task. Details will be explained.

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    Independent dialer system Закінчився left

    we wish to have someone connect and configure our [увійдіть, щоб побачити URL] to our SIP and Trunk

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    I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.

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    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    Voip Calling App -- 2 Закінчився left

    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

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    voip语音网关模块 Закінчився left

    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    Voip Calling App Закінчився left

    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

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    I need to create and confing server. i need to forward calls to it and to see the header and make tests can be done on linuxcpanelwinvps

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    Rebranding of Linphone Закінчився left

    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

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    Ringless Voicemail Закінчився left

    ...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] https://www

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    Tech Blogger -Telecommunications Закінчився left

    Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

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    La RCSOft software specializzata da anni in sviluppo software gestionali, per potere integrare al suo interno un CRM richiede una interfaccia con il Centralino Voip 3CX v15 in poi, per potere effettuare delle telefonate in uscita e per potere recuperare il numero chiamante in entrata in modo da posizionarsi sul form del cliente. L'interfaccia deve

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    Full Actionscript 3.0 SIP-RTMP bridge Закінчився left

    We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

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    Design project Закінчився left

    I’m throwing a sip and paint party. I’m looking for a artist to guide the class, to do a painting

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    webrtc asterisk Закінчився left

    Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

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    I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with

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    Dinstar GSM gateway VOIP Закінчився left

    I have installed Dinstar GSM gateway, but I have voice quality problem. Who can solve it, please contact with me

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    Actionscript 3.0 SIP client Закінчився left

    I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else

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    Voicemail Drop Windows Application Закінчився left

    ... The application will feature voicemail detection function where an outbound call is placed from a csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [увійдіть, щоб побачити URL] and sip.us. The application should

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